Hello:
I'm trying to register kphone to my ser proxy using authentication with user and password but I can't make it work. In the identity configuration of kphone I have set "jorge" in "user part of SIP URL" and "authentication username" and I have set "asterix" in "host part of SIP URL" and "outbound proxy" The same configuration with user and password works fine with windows messenger 4.7
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The error messages I get from kphone are these:
SipClient: Sending: 10:09:30.743 -------------------------------- REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175:5062 CSeq: 928 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 1234450217@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175:5062;transport=udp;q=1.0;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Receiving message...
SipClient: Received: 10:09:30.755 --------------------------------- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.175:5062 CSeq: 928 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.fbe3 From: "jorge" sip:jorge@asterix Call-ID: 1234450217@192.168.2.175 WWW-Authenticate: Digest realm="asterix", nonce="3fcb14af87f1bd1110526a8d29ad258492f165bd" Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
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And my ser.cfg file is this:
alias="asterix" alias=192.168.2.176
sip_warning=no
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R)
fifo="/tmp/ser_fifo"
# Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so"
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
route{
if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; };
record_route();
if (loose_route()) { t_relay(); break; };
if (uri==myself) {
if (uri =~ "^sip:9[0-9]*@asterix") { log ("enviando al asterisk\n"); t_relay_to_udp("192.168.2.175", "5060"); break; };
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication if (!www_authorize("asterix", "subscriber")) { www_challenge("asterix", "0"); break; };
save("location"); break; };
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; };
if (!t_relay()) { sl_reply_error(); };
}
You missed the most interesting part of the network dump. Because after receiving the 401 from Ser kphone should send the REIGSTER again, this time with Authentication header included. If kphone does not send such a request, then start your kphone from a terminal and watch at the debug output, maybe it gives you a hint.
Regards Nils On Monday 01 December 2003 14:38, Jorge Merlino wrote:
Hello:
I'm trying to register kphone to my ser proxy using authentication with user and password but I can't make it work. In the identity configuration of kphone I have set "jorge" in "user part of SIP URL" and "authentication username" and I have set "asterix" in "host part of SIP URL" and "outbound proxy" The same configuration with user and password works fine with windows messenger 4.7
The error messages I get from kphone are these:
SipClient: Sending: 10:09:30.743
REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175:5062 CSeq: 928 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 1234450217@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175:5062;transport=udp;q=1.0;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Receiving message...
SipClient: Received: 10:09:30.755
SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.175:5062 CSeq: 928 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.fbe3 From: "jorge" sip:jorge@asterix Call-ID: 1234450217@192.168.2.175 WWW-Authenticate: Digest realm="asterix", nonce="3fcb14af87f1bd1110526a8d29ad258492f165bd" Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
And my ser.cfg file is this:
alias="asterix" alias=192.168.2.176 sip_warning=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo"# Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (uri =~ "^sip:9[0-9]*@asterix") { log ("enviando al asterisk\n"); t_relay_to_udp("192.168.2.175", "5060"); break; }; if (method=="REGISTER") {# Uncomment this if you want to use digest authentication if (!www_authorize("asterix", "subscriber")) { www_challenge("asterix", "0"); break; };
save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (!t_relay()) { sl_reply_error(); }; }
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Nils:
I'm afraid there is no more interesting dump to send you, kphone keeps sending the same register message and SER keeps answering the same. By the way yesterday I tried to register kphone without authentication configured in SER and it didn't work either. Kphone keeps trying again and again although it reiceives an OK answer from SER every time. Oddly I've tested kphone with other proxies and always worked fine. Here is a dump of this other situation (by the way I'm using kphone v. 3.12):
SipClient: Listening UDP on port: 5060 SipClient: Our address: 192.168.2.175 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 09:00:38.025 -------------------------------- REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.176:5060' SipClient: Receiving message...
SipClient: Received: 09:00:38.041 --------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Contact: sip:jorge@192.168.2.175:5062;transport=udp;q=0.00;expires=337 Contact: sip:jorge@192.168.2.175;transport=udp;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
SipTransaction: Retransmit 1 (4000)
SipClient: Sending: 09:00:42.026 -------------------------------- REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Receiving message...
SipClient: Received: 09:00:42.038 --------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Contact: sip:jorge@192.168.2.175:5062;transport=udp;q=0.00;expires=333 Contact: sip:jorge@192.168.2.175;transport=udp;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
-----------------------------------------------------------------------------------------------------
Here is an example of a successful connection with other proxy (asterisk in this case)
SipClient: Listening UDP on port: 5062 SipClient: Our address: 192.168.2.176 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 08:47:08.026 -------------------------------- REGISTER sip:192.168.2.175 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.176:5062;rport CSeq: 1124 REGISTER To: "Luis Vazquez" sip:101@192.168.2.175 Expires: 600 From: "Luis Vazquez" sip:101@192.168.2.175 Call-ID: 546198833@192.168.2.176 Content-Length: 0 User-Agent: KPhone/3.14 Event: registration Allow-Events: presence Contact: "Luis Vazquez" sip:luis@192.168.2.176:5062;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.175:5060' SipClient: Receiving message...
SipClient: Received: 08:47:08.048 --------------------------------- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" sip:101@192.168.2.175 To: "Luis Vazquez" sip:101@192.168.2.175;tag=as3f7be343 Call-ID: 546198833@192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:101@192.168.2.175 Content-Length: 0
SipCall: Incoming response SipTransaction: Incoming Response SipClient: Receiving message...
SipClient: Received: 08:47:08.051 --------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" sip:101@192.168.2.175 To: "Luis Vazquez" sip:101@192.168.2.175;tag=as3f7be343 Call-ID: 546198833@192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 600 Contact: sip:101@192.168.2.175;expires=600 Date: Tue, 02 Dec 2003 11:46:29 GMT Content-Length: 0
SipCall: Incoming response SipCall: Checking for Contact and Record-Route SipCall: Setting Contact for this Call Member SipTransaction: Incoming Response ReRegistrationTimer (ms): 540000
Regards Jorge
Hello,
the unsucessfull test was made with kphone 3.12 and the sucessfull one with 3.14. Try to upgrade the older version to 3.14 as well. Also try to configure it the same way (i.e. use IP instead of 'asterix' in the configuration).
Jan.
On 02-12 09:05, Jorge Merlino wrote:
Nils:
I'm afraid there is no more interesting dump to send you, kphone keeps sending the same register message and SER keeps answering the same. By the way yesterday I tried to register kphone without authentication configured in SER and it didn't work either. Kphone keeps trying again and again although it reiceives an OK answer from SER every time. Oddly I've tested kphone with other proxies and always worked fine. Here is a dump of this other situation (by the way I'm using kphone v. 3.12):
SipClient: Listening UDP on port: 5060 SipClient: Our address: 192.168.2.175 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 09:00:38.025
REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.176:5060' SipClient: Receiving message...
SipClient: Received: 09:00:38.041
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Contact: sip:jorge@192.168.2.175:5062;transport=udp;q=0.00;expires=337 Contact: sip:jorge@192.168.2.175;transport=udp;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
SipTransaction: Retransmit 1 (4000)
SipClient: Sending: 09:00:42.026
REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Receiving message...
SipClient: Received: 09:00:42.038
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Contact: sip:jorge@192.168.2.175:5062;transport=udp;q=0.00;expires=333 Contact: sip:jorge@192.168.2.175;transport=udp;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
Here is an example of a successful connection with other proxy (asterisk in this case)
SipClient: Listening UDP on port: 5062 SipClient: Our address: 192.168.2.176 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 08:47:08.026
REGISTER sip:192.168.2.175 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.176:5062;rport CSeq: 1124 REGISTER To: "Luis Vazquez" sip:101@192.168.2.175 Expires: 600 From: "Luis Vazquez" sip:101@192.168.2.175 Call-ID: 546198833@192.168.2.176 Content-Length: 0 User-Agent: KPhone/3.14 Event: registration Allow-Events: presence Contact: "Luis Vazquez" sip:luis@192.168.2.176:5062;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.175:5060' SipClient: Receiving message...
SipClient: Received: 08:47:08.048
SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" sip:101@192.168.2.175 To: "Luis Vazquez" sip:101@192.168.2.175;tag=as3f7be343 Call-ID: 546198833@192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:101@192.168.2.175 Content-Length: 0
SipCall: Incoming response SipTransaction: Incoming Response SipClient: Receiving message...
SipClient: Received: 08:47:08.051
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" sip:101@192.168.2.175 To: "Luis Vazquez" sip:101@192.168.2.175;tag=as3f7be343 Call-ID: 546198833@192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 600 Contact: sip:101@192.168.2.175;expires=600 Date: Tue, 02 Dec 2003 11:46:29 GMT Content-Length: 0
SipCall: Incoming response SipCall: Checking for Contact and Record-Route SipCall: Setting Contact for this Call Member SipTransaction: Incoming Response ReRegistrationTimer (ms): 540000
Regards Jorge
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hello,
There is no difference in the version. The successful test was to show the differences and the client was registering to an asterix PBX. Both versions work fine with asterisk, none work with ser.
Regards, Jorge
On Wednesday 03 December 2003 13:41, Jan Janak wrote:
Hello,
the unsucessfull test was made with kphone 3.12 and the sucessfull one with 3.14. Try to upgrade the older version to 3.14 as well. Also try to configure it the same way (i.e. use IP instead of 'asterix' in the configuration).
Jan.
On 02-12 09:05, Jorge Merlino wrote:
Nils:
I'm afraid there is no more interesting dump to send you, kphone keeps sending the same register message and SER keeps answering the same. By the way yesterday I tried to register kphone without authentication configured in SER and it didn't work either. Kphone keeps trying again and again although it reiceives an OK answer from SER every time. Oddly I've tested kphone with other proxies and always worked fine. Here is a dump of this other situation (by the way I'm using kphone v. 3.12):
SipClient: Listening UDP on port: 5060 SipClient: Our address: 192.168.2.175 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 09:00:38.025
REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.176:5060' SipClient: Receiving message...
SipClient: Received: 09:00:38.041
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Contact: sip:jorge@192.168.2.175:5062;transport=udp;q=0.00;expires=337 Contact: sip:jorge@192.168.2.175;transport=udp;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
SipTransaction: Retransmit 1 (4000)
SipClient: Sending: 09:00:42.026
REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Receiving message...
SipClient: Received: 09:00:42.038
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Contact: sip:jorge@192.168.2.175:5062;transport=udp;q=0.00;expires=333 Contact: sip:jorge@192.168.2.175;transport=udp;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
Here is an example of a successful connection with other proxy (asterisk in this case)
SipClient: Listening UDP on port: 5062 SipClient: Our address: 192.168.2.176 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 08:47:08.026
REGISTER sip:192.168.2.175 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.176:5062;rport CSeq: 1124 REGISTER To: "Luis Vazquez" sip:101@192.168.2.175 Expires: 600 From: "Luis Vazquez" sip:101@192.168.2.175 Call-ID: 546198833@192.168.2.176 Content-Length: 0 User-Agent: KPhone/3.14 Event: registration Allow-Events: presence Contact: "Luis Vazquez" sip:luis@192.168.2.176:5062;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.175:5060' SipClient: Receiving message...
SipClient: Received: 08:47:08.048
SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" sip:101@192.168.2.175 To: "Luis Vazquez" sip:101@192.168.2.175;tag=as3f7be343 Call-ID: 546198833@192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:101@192.168.2.175 Content-Length: 0
SipCall: Incoming response SipTransaction: Incoming Response SipClient: Receiving message...
SipClient: Received: 08:47:08.051
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" sip:101@192.168.2.175 To: "Luis Vazquez" sip:101@192.168.2.175;tag=as3f7be343 Call-ID: 546198833@192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 600 Contact: sip:101@192.168.2.175;expires=600 Date: Tue, 02 Dec 2003 11:46:29 GMT Content-Length: 0
SipCall: Incoming response SipCall: Checking for Contact and Record-Route SipCall: Setting Contact for this Call Member SipTransaction: Incoming Response ReRegistrationTimer (ms): 540000
Regards Jorge
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
I don't know, I am using kphone 3.14 and it works fine with ser.
Jan.
On 04-12 12:09, Jorge Merlino wrote:
Hello,
There is no difference in the version. The successful test was to show the differences and the client was registering to an asterix PBX. Both versions work fine with asterisk, none work with ser.
Regards, Jorge
On Wednesday 03 December 2003 13:41, Jan Janak wrote:
Hello,
the unsucessfull test was made with kphone 3.12 and the sucessfull one with 3.14. Try to upgrade the older version to 3.14 as well. Also try to configure it the same way (i.e. use IP instead of 'asterix' in the configuration).
Jan.
On 02-12 09:05, Jorge Merlino wrote:
Nils:
I'm afraid there is no more interesting dump to send you, kphone keeps sending the same register message and SER keeps answering the same. By the way yesterday I tried to register kphone without authentication configured in SER and it didn't work either. Kphone keeps trying again and again although it reiceives an OK answer from SER every time. Oddly I've tested kphone with other proxies and always worked fine. Here is a dump of this other situation (by the way I'm using kphone v. 3.12):
SipClient: Listening UDP on port: 5060 SipClient: Our address: 192.168.2.175 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 09:00:38.025
REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.176:5060' SipClient: Receiving message...
SipClient: Received: 09:00:38.041
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Contact: sip:jorge@192.168.2.175:5062;transport=udp;q=0.00;expires=337 Contact: sip:jorge@192.168.2.175;transport=udp;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
SipTransaction: Retransmit 1 (4000)
SipClient: Sending: 09:00:42.026
REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix Expires: 900 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" sip:jorge@192.168.2.175;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Receiving message...
SipClient: Received: 09:00:42.038
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" sip:jorge@asterix;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" sip:jorge@asterix Call-ID: 123630670@192.168.2.175 Contact: sip:jorge@192.168.2.175:5062;transport=udp;q=0.00;expires=333 Contact: sip:jorge@192.168.2.175;transport=udp;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0
Here is an example of a successful connection with other proxy (asterisk in this case)
SipClient: Listening UDP on port: 5062 SipClient: Our address: 192.168.2.176 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)'
SipClient: Sending: 08:47:08.026
REGISTER sip:192.168.2.175 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.176:5062;rport CSeq: 1124 REGISTER To: "Luis Vazquez" sip:101@192.168.2.175 Expires: 600 From: "Luis Vazquez" sip:101@192.168.2.175 Call-ID: 546198833@192.168.2.176 Content-Length: 0 User-Agent: KPhone/3.14 Event: registration Allow-Events: presence Contact: "Luis Vazquez" sip:luis@192.168.2.176:5062;transport=udp;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Sending to '192.168.2.175:5060' SipClient: Receiving message...
SipClient: Received: 08:47:08.048
SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" sip:101@192.168.2.175 To: "Luis Vazquez" sip:101@192.168.2.175;tag=as3f7be343 Call-ID: 546198833@192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:101@192.168.2.175 Content-Length: 0
SipCall: Incoming response SipTransaction: Incoming Response SipClient: Receiving message...
SipClient: Received: 08:47:08.051
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" sip:101@192.168.2.175 To: "Luis Vazquez" sip:101@192.168.2.175;tag=as3f7be343 Call-ID: 546198833@192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 600 Contact: sip:101@192.168.2.175;expires=600 Date: Tue, 02 Dec 2003 11:46:29 GMT Content-Length: 0
SipCall: Incoming response SipCall: Checking for Contact and Record-Route SipCall: Setting Contact for this Call Member SipTransaction: Incoming Response ReRegistrationTimer (ms): 540000
Regards Jorge
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
I solved the problem at last; it seems the cause is, as you told me, that kphone cannot handle host names (at least when they are not fully qualified ones) in the host part of the sip URL. It worked just fine when I used the IP number.
Thank you for your time.
Regards, Jorge
On Thursday 04 December 2003 16:55, Jan Janak wrote:
I don't know, I am using kphone 3.14 and it works fine with ser.
Jan.
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Jan Janak -
Jorge Merlino -
Nils Ohlmeier